Re: [asterisk-users] Where to download Junghanns ISDNguard software?

Nick Richardson a écrit : Hi list, I recently purchased an ISDNguard from Junghanns. It came with no software and there is no sign on their website or in any of their documentation where to download it. I have looked in http://www.junghanns.net/downloads/ and there is no sign of it there

Re: [asterisk-users] Where to download Junghanns ISDNguard software?

Hi Nick, For using ISDNGuard, you can using res_watchdog from the bristuff patch. I attach for you a version who have patched and work for asterisk 1.2.24. It's an archive with a makefile with just res_watchdog. We have some clients who using this version from many month ago. Sylvain Le jeudi

Re: [asterisk-users] tdm fxo module always busy

Hi Chawki, it is not uncommon that FXO or FXS modules do not work even if no error message is shown when wctdm module is loaded. Have you tried to replace/swap your FXO modules? Giorgio chawki hammoud wrote: Hi: I have a digium tdm04B. One fxo module always gives a faulty busy signal when

[asterisk-users] Friday VOIP Users Conference 12:30 PM EDT

This Friday at 12:30 PM EDT We hope to hear more about Astricon and the 1.6 version. A UK legal professional, John Halton of Cripps Harries Hall LLP, joins us to discuss how the law is coming to terms with VOIP. We also expect a visit from Arick of IPKall about what's cooking with them. Most of

[asterisk-users] Multi tenant

Hi all, i just wanted to know if any one has done any multi-tenant version of the asterisk. thanks ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit:

Re: [asterisk-users] tdm fxo module always busy

On 10/4/07, chawki hammoud [EMAIL PROTECTED] wrote: I have a digium tdm04B. One fxo module always gives a faulty busy signal when a call comes in. I swaped places and re-compiled zaptel with no change. Is this a faulty fxo or configuration issue. Hi, I suggest you post your related zaptel

Re: [asterisk-users] Infuriating problems: no dial tone, dropped calls, no voice: 1.2.13 and 1.4.11

Christian Weeks wrote: Hi I've had an asterisk setup for the past 15 months, based on the debian asterisk packaging. Until late August of this year, I had no problems once initial setup was complete- the system worked essentially flawlessly. Since August I have been having exceedingly

Re: [asterisk-users] extensions.conf vs. AEL

All this discussion is pointless. As pointless as the discussion of assembly versus high-level languages decades ago. Except most people rooting for extension.conf don't even have the technical and conceptual amplitude to understand what they are talking about... they just want some telephony

Re: [asterisk-users] No audio on Zap (T1/PRI) channels

Steve Edwards wrote: On Wed, 3 Oct 2007, Steve Totaro wrote: Kevin P. Fleming wrote: Steve Edwards wrote: [trunkgroups] trunkgroup = 1,24,96 spanmap = 1,1,0 spanmap = 2,1,2

[asterisk-users] Secondary Dialtone and selecting a specific line from Zap/g

Dear Mojo; That is primary fine, but there are two issues looking for help about them: 1) Based on your below example (dialing *4*18005551212 to select channel 4), the question is how to give second dial tone just after dialing the *4* (indicating the channel was captured)? 2) How to let this

[asterisk-users] Secondary Dialtone and selecting a specific line from Zap/g

Dear Walt; Maybe I did not understand any thing from below :) - Are you talking about configuration to be done on the Telephone device is self or on the AVAYA server it self? If it is on the telephone device, so how you will give a second dial tone and you do not know if there is available

Re: [asterisk-users] Configuration files inside SQLite3

Hi Mark, Am Mittwoch, den 03.10.2007, 11:15 -0500 schrieb Mark Michelson: GNUbie wrote: Hello all, Is it possible to store, read and write configuration files in an SQLite3 database instead of using the configuration files inside the /etc/asterisk/ directory? If it is then can you

[asterisk-users] About Megaco

Hi all, I've been searching for a while and haven't found if asterisk supports already or if it's going to support h.248. thanks Eve ¡Sé un mejor ambientalista! Encuentra consejos para cuidar el

Re: [asterisk-users] About Megaco

On Oct 4, 2007, at 8:39 AM, Steve Totaro wrote: Try searching using MGCP which is what Megaco evolved into. http://www.voip-info.org/wiki-Asterisk+MGCP+channels Thanks, Steve Totaro Too bad the MGCP channel isn't the full implementation. /b ___

Re: [asterisk-users] extensions.conf vs. AEL

On Thursday 04 October 2007 07:07:47 Barzilai Spinak wrote: All this discussion is pointless. As pointless as the discussion of assembly versus high-level languages decades ago. As one of the main architects, I don't find this discussion pointless. My personal opinion of AEL is that it's

[asterisk-users] Line State

Hello all, I need a little help to check the state of the line from asterisk on aa TDM400P because when the telco lines goes down, asterisk get that line for outgoing calls. There is a way to check it out? And when all lines are busy to do outgoing calls how can i do to callback the people

Re: [asterisk-users] About Megaco

Floyd wrote: Hi all, I've been searching for a while and haven't found if asterisk supports already or if it's going to support h.248. thanks Eve Try searching using MGCP which is what Megaco evolved into. http://www.voip-info.org/wiki-Asterisk+MGCP+channels Thanks, Steve

Re: [asterisk-users] Multi tenant

Hi Mujtaba, We have a multi-tenant version of our Asterisk based management and end-user software called Thirdlane PBX Manager. You can see a demo of a single-tenant version on our web site http://www.thirdlane.com/pbxmanager.htm the multi-tenant adds tenant and DID management, and allows to

[asterisk-users] Fwd: [asterisk-dev] chan_h323 and chan_oh323 compatibilities

-- Forwarded message -- From: Tzafrir Cohen [EMAIL PROTECTED] Date: Oct 4, 2007 12:56 PM Subject: Re: [asterisk-dev] chan_h323 and chan_oh323 compatibilities To: [EMAIL PROTECTED] Hi On Thu, Oct 04, 2007 at 11:46:30AM -0300, Caciano Machado wrote: I'm receiving a lot of warning

[asterisk-users] Asterisk Caller ID Info

Hi Asterisk Users, I was wondering why a call that is received from Asterisk shows a caller ID 'Unknown' . So here is the scenario, 'A' calls 'Asterisk'. 'A' joins a conference on 'Asterisk'. 'Asterisk' calls 'B'. 'B' gets joined to the same conference also. 'B' somehow receives the caller ID

[asterisk-users] Dial-Chain interrupted by Operator Called Party not reachable Messages

Hi, I have the following problem: I want asterisk to dial a chain of n-numbers until somebody picks up the line. I am using Digium E1 Hardware (zaptel) for dialing out. Dialing a Chain is basically no problem, I use somwthing like: dial(no1,50) dial(no2,50) dial(no3,50) However, If no1 is not

Re: [asterisk-users] Using PHP to reload extensions

On Wed, Oct 03, 2007 at 09:10:58PM -0500, Moises Silva wrote: If you are running the script from a web server, the script gets executed with the web server process permissions, hence, probably does not have access to /var/run/asterisk.ctl. You can give permissions to your web server, or

[asterisk-users] need suggestion

Dear All, my client wants a asterisk pbx with 30 FXO 30 FXS analogue ports, please suggest if sangoma A400 is a good option for that. Also please suggest server hardware. regards, Umair ___ --Bandwidth and Colocation Provided by

Re: [asterisk-users] Supermicro PDSME+ and TE110P [ ref:00D36mPe.50033qy57:ref ] NEW CASE 22828

I never thought very useful to search in this http://www.digium.com/en/docs/misc/compatibility_notes.php page, as it looked rather static for holding such compatibility issues. You proved me I was wrong. Taking this e1000 driver issue as an example, this http://sourceforge.net/projects/e1000

[asterisk-users] Question FXO Port

Hello list I am new in this list. Before I wrote this email, i search with google and in the list arichves for the question. I look for a possibility to install FXO ports not over RJ11 Ports. I will install the Ports by LSA+ Patch panel. Someone an idea ore link? Thanks for help. Bye MZ PS:

[asterisk-users] Setting caller id value on outgoing calls using .call files

Hi all, I was looking at a way to add the caller id to the outgoing calls (which are made using .call files) using asterisk. Any ideas how to do this ? Currently I get 'Unknown' number displayed on my phone when asterisk makes an outgoing call. Also using something like this is not working as it

Re: [asterisk-users] Setting caller id value on outgoing calls using .call files

Thanks guys. No need to reply. I got my answer from someone. On 10/4/07, Arpit Mehta [EMAIL PROTECTED] wrote: Hi all, I was looking at a way to add the caller id to the outgoing calls (which are made using .call files) using asterisk. Any ideas how to do this ? Currently I get 'Unknown'

Re: [asterisk-users] Asterisk Caller ID Info

Thanks a lot guys. I got my answer from someone. :) On 10/4/07, Arpit Mehta [EMAIL PROTECTED] wrote: Also what are the ways if any to set this DNIS or RDNIS information ? Regards Arpit On 10/4/07, Arpit Mehta [EMAIL PROTECTED] wrote: Hi Asterisk Users, I was wondering why a

Re: [asterisk-users] Asterisk Caller ID Info

Also what are the ways if any to set this DNIS or RDNIS information ? Regards Arpit On 10/4/07, Arpit Mehta [EMAIL PROTECTED] wrote: Hi Asterisk Users, I was wondering why a call that is received from Asterisk shows a caller ID 'Unknown' . So here is the scenario, 'A' calls 'Asterisk'.

[asterisk-users] Voicemail/dtmf not working?

Hi, I am setting up an asterisk server for testing purposes and cannot get voicemail to work at all. My host OS is Linux From Scratch 6.3 and the asterisk software versions I built are zaptel-1.4.5.1 and asterisk-1.4.12. I am using the Ekiga softphone on my Ubuntu desktop machine. My asterisk

Re: [asterisk-users] Cisco 7940G licensing with asterisk

There are other Gigabit SIP phones from Nortel and Avaya, if my memory serves me right. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit:

Re: [asterisk-users] Asterisk 1.4.12 and Asterisk-addons 1.4.3 released

Just forget it about the 1.2 mantra, it's not going to happen. Focus your energy elsewhere. Lot's of bug fixes are good. Even Cisco comes out with regular bug fixes for IOS. Open source just makes things more visible. -Original Message- From: Steve Totaro [mailto:[EMAIL PROTECTED]

Re: [asterisk-users] G.722: ast_channel_make_compatible failure

Well I know. My problem is, that the phone offering g722 could do alaw as well. I expected asterisk should just chose alaw for the communication - no transcoding is necessary then... Please help. Thanks, Ondrej Kevin P. Fleming wrote: Ondrej Valousek wrote: [Sep 20 10:14:32]

Re: [asterisk-users] Setting caller id value on outgoing calls using .call files

Arpit Mehta wrote: I was looking at a way to add the caller id to the outgoing calls (which are made using .call files) using asterisk. Any ideas how to do this ? Add Callerid: Name 123 to the call file. Regards, Philipp Kempgen -- amooma GmbH - Bachstr. 126 - 56566 Neuwied -

Re: [asterisk-users] Voicemail/dtmf not working?

Alan Lord wrote: sip.conf = [100] type=friend callerid=Alan Lord secret=** qualify=yes ; Qualify peer is no more than 2000 ms away nat=no ; This phone is not natted host=dynamic ; This device registers with us canreinvite=no ; Asterisk

Re: [asterisk-users] Setting caller id value on outgoing calls using .call files

On 10/4/07, Arpit Mehta [EMAIL PROTECTED] wrote: I was looking at a way to add the caller id to the outgoing calls (which are made using .call files) using asterisk. Any ideas how to do this ? Currently I get 'Unknown' number displayed on my phone when asterisk makes an outgoing call. Add a

Re: [asterisk-users] Setting caller id value on outgoing calls using .call files

On Thu, 2007-10-04 at 16:10 -0400, Arpit Mehta wrote: exten = _.,4,Set(CALLERID(all)=Joe 911) exten = _.,5,system(cp /var/spool/asterisk/1.call /var/spool/asterisk/outgoing/) You need to set the Caller ID in the call file itself. The sample call file (sample.call) in the Asterisk source

Re: [asterisk-users] Asterisk Caller ID Info

Arpit Mehta wrote: Thanks a lot guys. I got my answer from someone. :) Not that I'm interested in this specific issue, but usually it won't hurt to share your solution with the other list members (and archives). Doesn't apply to this case maybe. Cheers, Philipp Kempgen -- amooma GmbH -

[asterisk-users] #modprobe wctdm or #modprobe zaptel

Hi list; I need to run the command modprobe wctdm and whenever I write it, then it gives me the following message: FATAL: Module wctdm not found FATAL: Error running install command for wctdm So, do I have to run that command from specific path? Or what is the problem? Any help? Regards

Re: [asterisk-users] Using PHP to reload extensions

No, because then asterisk would be presented three arguments: '-rx', 'extensions', and 'reload' -- as 'extensions' is not a command by itself, and the 'reload' appears superfluous to asterisk, this would not work as desired. Asterisk needs to be presented two arguments - the first is '-rx',

Re: [asterisk-users] Voicemail/dtmf not working?

At 11:10 10/4/2007, Alan Lord wrote: Hi, I am setting up an asterisk server for testing purposes and cannot get voicemail to work at all. My host OS is Linux From Scratch 6.3 and the asterisk software versions I built are zaptel-1.4.5.1 and asterisk-1.4.12. I am using the Ekiga

Re: [asterisk-users] G.722: ast_channel_make_compatible failure

Ondrej Valousek wrote: My problem is, that the phone offering g722 could do alaw as well. I expected asterisk should just chose alaw for the communication - no transcoding is necessary then... That is not how Asterisk works, and is well known in the community as something that users would

[asterisk-users] Strange Call problems on some numbers

I am having some really strange problems calling from 2 asterisk boxes of mine. One is version 1.2.22 the other 1.2.18. The problem is identical on both boxes. When I try to call certain numbers (8006375410, for instance) the call rings and rings and rings. Eventually the receiving end will pick

Re: [asterisk-users] Strange Call problems on some numbers

Mail Lists wrote: I am having some really strange problems calling from 2 asterisk boxes of mine. One is version 1.2.22 the other 1.2.18. The problem is identical on both boxes. When I try to call certain numbers (8006375410, for instance) the call rings and rings and rings. Eventually the

Re: [asterisk-users] #modprobe wctdm or #modprobe zaptel

bilal ghayyad wrote: Hi list; I need to run the command modprobe wctdm and whenever I write it, then it gives me the following message: FATAL: Module wctdm not found FATAL: Error running install command for wctdm So, do I have to run that command from specific path? Or what is the

Re: [asterisk-users] Dial-Chain interrupted by Operator Called Party not reachable Messages

Christoph Adomeit wrote: Hi, I have the following problem: I want asterisk to dial a chain of n-numbers until somebody picks up the line. I am using Digium E1 Hardware (zaptel) for dialing out. Dialing a Chain is basically no problem, I use somwthing like: dial(no1,50) dial(no2,50)

Re: [asterisk-users] Secondary Dialtone and selecting a specific line from Zap/g

The problem here, if it's a problem, is that it IS a POTS dialtone and not asterisk's. So you can dial only what the telco lets you dial. Al lists wrote: Here is how i overcome this problem, ignorpat = 9 exten = 9*,1,Dial(ZAP/1/w) press 9* from your handset and after 1 second you have

Re: [asterisk-users] Strange Call problems on some numbers

Also, so TFN's do not answer the line. Airline TFN are famous for doing this. Arick Davis -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro Sent: Thursday, October 04, 2007 2:53 PM To: Asterisk Users Mailing List - Non-Commercial Discussion

Re: [asterisk-users] Setting caller id value on outgoing calls using .call files

On Thu, Oct 04, 2007 at 04:10:09PM -0400, Arpit Mehta wrote: Hi all, I was looking at a way to add the caller id to the outgoing calls (which are made using .call files) using asterisk. Any ideas how to do this ? Currently I get 'Unknown' number displayed on my phone when asterisk makes an

Re: [asterisk-users] Using PHP to reload extensions

Hello, I am trying to use PHP to reload the extensions in an Asterisk installation. I keep getting this error: Easiest way without compromising security or changing permissions. Use the AMI. 1. Download phpagi (Just google it) 2. Use it to connect to the Manager interface 3. Use it to

Re: [asterisk-users] Where to download Junghanns ISDNguard software?

What if you don't use or want to use bristuff? We use Digium PRI cards and don't need any of the BRIstuff On 10/4/07, Sylvain Boily [EMAIL PROTECTED] wrote: Hi Nick, For using ISDNGuard, you can using res_watchdog from the bristuff patch. I attach for you a version who have patched and

Re: [asterisk-users] No audio on Zap (T1/PRI) channels

Steve Totaro wrote: Steve Edwards wrote: I have 12 T1's going into 3 servers, 4 in each into Digium, Inc. Wildcard TE410P Quad-Span togglable E1/T1/J1 card 3.3v (rev 02) cards. Each group of T1's have the primary D on 24 and the secondary D on 96. The first server (ts20) and the last

[asterisk-users] Asterisk status in Debian

Hello, This is a update on the current status of Asterisk in Debian. Apologies for the really long mail, it is targetted both to users and maintainers :) I'm Ccing asterisk-users as a one-time thing; users that are interested can subscribe to our list[1] for updates to prevent noise on a

Re: [asterisk-users] Where to download Junghanns ISDNguard software?

Le vendredi 05 octobre 2007 à 12:13 +1000, Nick Richardson a écrit : What if you don't use or want to use bristuff? We use Digium PRI cards and don't need any of the BRIstuff yep i know ... but res_watchdog is part of bristuff. My tarball have just the res_watchdog module with a makefile.